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286 lines
8.4 KiB
C
286 lines
8.4 KiB
C
/* GStreamer
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* Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/**
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* SECTION:element-rtpL16depay
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* @see_also: rtpL16pay
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*
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* Extract raw audio from RTP packets according to RFC 3551.
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* For detailed information see: http://www.rfc-editor.org/rfc/rfc3551.txt
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*
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* <refsect2>
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* <title>Example pipeline</title>
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* |[
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* gst-launch-1.0 udpsrc caps='application/x-rtp, media=(string)audio, clock-rate=(int)44100, encoding-name=(string)L16, encoding-params=(string)1, channels=(int)1, payload=(int)96' ! rtpL16depay ! pulsesink
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* ]| This example pipeline will depayload an RTP raw audio stream. Refer to
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* the rtpL16pay example to create the RTP stream.
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* </refsect2>
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <string.h>
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#include <stdlib.h>
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#include <gst/audio/audio.h>
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#include "gstrtpL16depay.h"
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#include "gstrtpchannels.h"
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#include "gstrtputils.h"
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GST_DEBUG_CATEGORY_STATIC (rtpL16depay_debug);
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#define GST_CAT_DEFAULT (rtpL16depay_debug)
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static GstStaticPadTemplate gst_rtp_L16_depay_src_template =
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GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-raw, "
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"format = (string) S16BE, "
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"layout = (string) interleaved, "
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"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
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);
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static GstStaticPadTemplate gst_rtp_L16_depay_sink_template =
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GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("application/x-rtp, "
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"media = (string) \"audio\", " "clock-rate = (int) [ 1, MAX ], "
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/* "channels = (int) [1, MAX]" */
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/* "emphasis = (string) ANY" */
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/* "channel-order = (string) ANY" */
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"encoding-name = (string) \"L16\";"
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"application/x-rtp, "
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"media = (string) \"audio\", "
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"payload = (int) { " GST_RTP_PAYLOAD_L16_STEREO_STRING ", "
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GST_RTP_PAYLOAD_L16_MONO_STRING " }," "clock-rate = (int) [ 1, MAX ]"
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/* "channels = (int) [1, MAX]" */
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/* "emphasis = (string) ANY" */
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/* "channel-order = (string) ANY" */
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)
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);
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#define gst_rtp_L16_depay_parent_class parent_class
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G_DEFINE_TYPE (GstRtpL16Depay, gst_rtp_L16_depay, GST_TYPE_RTP_BASE_DEPAYLOAD);
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static gboolean gst_rtp_L16_depay_setcaps (GstRTPBaseDepayload * depayload,
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GstCaps * caps);
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static GstBuffer *gst_rtp_L16_depay_process (GstRTPBaseDepayload * depayload,
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GstRTPBuffer * rtp);
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static void
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gst_rtp_L16_depay_class_init (GstRtpL16DepayClass * klass)
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{
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GstElementClass *gstelement_class;
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GstRTPBaseDepayloadClass *gstrtpbasedepayload_class;
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gstelement_class = (GstElementClass *) klass;
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gstrtpbasedepayload_class = (GstRTPBaseDepayloadClass *) klass;
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gstrtpbasedepayload_class->set_caps = gst_rtp_L16_depay_setcaps;
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gstrtpbasedepayload_class->process_rtp_packet = gst_rtp_L16_depay_process;
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gst_element_class_add_static_pad_template (gstelement_class,
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&gst_rtp_L16_depay_src_template);
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gst_element_class_add_static_pad_template (gstelement_class,
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&gst_rtp_L16_depay_sink_template);
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gst_element_class_set_static_metadata (gstelement_class,
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"RTP audio depayloader", "Codec/Depayloader/Network/RTP",
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"Extracts raw audio from RTP packets",
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"Zeeshan Ali <zak147@yahoo.com>," "Wim Taymans <wim.taymans@gmail.com>");
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GST_DEBUG_CATEGORY_INIT (rtpL16depay_debug, "rtpL16depay", 0,
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"Raw Audio RTP Depayloader");
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}
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static void
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gst_rtp_L16_depay_init (GstRtpL16Depay * rtpL16depay)
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{
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}
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static gint
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gst_rtp_L16_depay_parse_int (GstStructure * structure, const gchar * field,
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gint def)
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{
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const gchar *str;
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gint res;
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if ((str = gst_structure_get_string (structure, field)))
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return atoi (str);
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if (gst_structure_get_int (structure, field, &res))
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return res;
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return def;
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}
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static gboolean
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gst_rtp_L16_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
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{
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GstStructure *structure;
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GstRtpL16Depay *rtpL16depay;
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gint clock_rate, payload;
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gint channels;
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GstCaps *srccaps;
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gboolean res;
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const gchar *channel_order;
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const GstRTPChannelOrder *order;
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GstAudioInfo *info;
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rtpL16depay = GST_RTP_L16_DEPAY (depayload);
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structure = gst_caps_get_structure (caps, 0);
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payload = 96;
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gst_structure_get_int (structure, "payload", &payload);
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switch (payload) {
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case GST_RTP_PAYLOAD_L16_STEREO:
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channels = 2;
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clock_rate = 44100;
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break;
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case GST_RTP_PAYLOAD_L16_MONO:
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channels = 1;
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clock_rate = 44100;
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break;
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default:
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/* no fixed mapping, we need clock-rate */
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channels = 0;
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clock_rate = 0;
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break;
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}
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/* caps can overwrite defaults */
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clock_rate =
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gst_rtp_L16_depay_parse_int (structure, "clock-rate", clock_rate);
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if (clock_rate == 0)
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goto no_clockrate;
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channels =
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gst_rtp_L16_depay_parse_int (structure, "encoding-params", channels);
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if (channels == 0) {
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channels = gst_rtp_L16_depay_parse_int (structure, "channels", channels);
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if (channels == 0) {
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/* channels defaults to 1 otherwise */
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channels = 1;
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}
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}
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depayload->clock_rate = clock_rate;
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info = &rtpL16depay->info;
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gst_audio_info_init (info);
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info->finfo = gst_audio_format_get_info (GST_AUDIO_FORMAT_S16BE);
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info->rate = clock_rate;
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info->channels = channels;
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info->bpf = (info->finfo->width / 8) * channels;
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/* add channel positions */
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channel_order = gst_structure_get_string (structure, "channel-order");
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order = gst_rtp_channels_get_by_order (channels, channel_order);
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rtpL16depay->order = order;
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if (order) {
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memcpy (info->position, order->pos,
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sizeof (GstAudioChannelPosition) * channels);
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gst_audio_channel_positions_to_valid_order (info->position, info->channels);
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} else {
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GST_ELEMENT_WARNING (rtpL16depay, STREAM, DECODE,
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(NULL), ("Unknown channel order '%s' for %d channels",
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GST_STR_NULL (channel_order), channels));
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/* create default NONE layout */
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gst_rtp_channels_create_default (channels, info->position);
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}
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srccaps = gst_audio_info_to_caps (info);
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res = gst_pad_set_caps (depayload->srcpad, srccaps);
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gst_caps_unref (srccaps);
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return res;
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/* ERRORS */
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no_clockrate:
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{
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GST_ERROR_OBJECT (depayload, "no clock-rate specified");
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return FALSE;
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}
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}
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static GstBuffer *
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gst_rtp_L16_depay_process (GstRTPBaseDepayload * depayload, GstRTPBuffer * rtp)
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{
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GstRtpL16Depay *rtpL16depay;
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GstBuffer *outbuf;
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gint payload_len;
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gboolean marker;
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rtpL16depay = GST_RTP_L16_DEPAY (depayload);
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payload_len = gst_rtp_buffer_get_payload_len (rtp);
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if (payload_len <= 0)
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goto empty_packet;
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GST_DEBUG_OBJECT (rtpL16depay, "got payload of %d bytes", payload_len);
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outbuf = gst_rtp_buffer_get_payload_buffer (rtp);
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marker = gst_rtp_buffer_get_marker (rtp);
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if (marker) {
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/* mark talk spurt with RESYNC */
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GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
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}
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outbuf = gst_buffer_make_writable (outbuf);
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if (rtpL16depay->order &&
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!gst_audio_buffer_reorder_channels (outbuf,
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rtpL16depay->info.finfo->format, rtpL16depay->info.channels,
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rtpL16depay->info.position, rtpL16depay->order->pos)) {
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goto reorder_failed;
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}
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gst_rtp_drop_meta (GST_ELEMENT_CAST (rtpL16depay), outbuf,
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g_quark_from_static_string (GST_META_TAG_AUDIO_STR));
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return outbuf;
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/* ERRORS */
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empty_packet:
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{
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GST_ELEMENT_WARNING (rtpL16depay, STREAM, DECODE,
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("Empty Payload."), (NULL));
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return NULL;
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}
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reorder_failed:
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{
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GST_ELEMENT_ERROR (rtpL16depay, STREAM, DECODE,
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("Channel reordering failed."), (NULL));
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return NULL;
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}
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}
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gboolean
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gst_rtp_L16_depay_plugin_init (GstPlugin * plugin)
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{
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return gst_element_register (plugin, "rtpL16depay",
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GST_RANK_SECONDARY, GST_TYPE_RTP_L16_DEPAY);
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}
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